libavcodec/libopusenc.c
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 /*
  * Opus encoder using libopus
  * Copyright (c) 2012 Nathan Caldwell
  *
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  * This file is part of FFmpeg.
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  *
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  * FFmpeg is free software; you can redistribute it and/or
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  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
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  * FFmpeg is distributed in the hope that it will be useful,
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  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
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  * License along with FFmpeg; if not, write to the Free Software
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  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #include <opus.h>
 #include <opus_multistream.h>
 
 #include "libavutil/opt.h"
 #include "avcodec.h"
 #include "bytestream.h"
 #include "internal.h"
 #include "libopus.h"
 #include "vorbis.h"
 #include "audio_frame_queue.h"
 
 typedef struct LibopusEncOpts {
     int vbr;
     int application;
     int packet_loss;
     int complexity;
     float frame_duration;
     int packet_size;
     int max_bandwidth;
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     int mapping_family;
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 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
     int apply_phase_inv;
 #endif
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 } LibopusEncOpts;
 
 typedef struct LibopusEncContext {
     AVClass *class;
     OpusMSEncoder *enc;
     int stream_count;
     uint8_t *samples;
     LibopusEncOpts opts;
     AudioFrameQueue afq;
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     const uint8_t *encoder_channel_map;
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 } LibopusEncContext;
 
 static const uint8_t opus_coupled_streams[8] = {
     0, 1, 1, 2, 2, 2, 2, 3
 };
 
 /* Opus internal to Vorbis channel order mapping written in the header */
 static const uint8_t opus_vorbis_channel_map[8][8] = {
     { 0 },
     { 0, 1 },
     { 0, 2, 1 },
     { 0, 1, 2, 3 },
     { 0, 4, 1, 2, 3 },
     { 0, 4, 1, 2, 3, 5 },
     { 0, 4, 1, 2, 3, 5, 6 },
     { 0, 6, 1, 2, 3, 4, 5, 7 },
 };
 
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 /* libavcodec to libopus channel order mapping, passed to libopus */
 static const uint8_t libavcodec_libopus_channel_map[8][8] = {
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     { 0 },
     { 0, 1 },
     { 0, 1, 2 },
     { 0, 1, 2, 3 },
     { 0, 1, 3, 4, 2 },
     { 0, 1, 4, 5, 2, 3 },
     { 0, 1, 5, 6, 2, 4, 3 },
     { 0, 1, 6, 7, 4, 5, 2, 3 },
 };
 
 static void libopus_write_header(AVCodecContext *avctx, int stream_count,
                                  int coupled_stream_count,
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                                  int mapping_family,
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                                  const uint8_t *channel_mapping)
 {
     uint8_t *p   = avctx->extradata;
     int channels = avctx->channels;
 
     bytestream_put_buffer(&p, "OpusHead", 8);
     bytestream_put_byte(&p, 1); /* Version */
     bytestream_put_byte(&p, channels);
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     bytestream_put_le16(&p, avctx->initial_padding); /* Lookahead samples at 48kHz */
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     bytestream_put_le32(&p, avctx->sample_rate); /* Original sample rate */
     bytestream_put_le16(&p, 0); /* Gain of 0dB is recommended. */
 
     /* Channel mapping */
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     bytestream_put_byte(&p, mapping_family);
     if (mapping_family != 0) {
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         bytestream_put_byte(&p, stream_count);
         bytestream_put_byte(&p, coupled_stream_count);
         bytestream_put_buffer(&p, channel_mapping, channels);
     }
 }
 
 static int libopus_configure_encoder(AVCodecContext *avctx, OpusMSEncoder *enc,
                                      LibopusEncOpts *opts)
 {
     int ret;
 
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     if (avctx->global_quality) {
         av_log(avctx, AV_LOG_ERROR,
                "Quality-based encoding not supported, "
                "please specify a bitrate and VBR setting.\n");
         return AVERROR(EINVAL);
     }
 
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     ret = opus_multistream_encoder_ctl(enc, OPUS_SET_BITRATE(avctx->bit_rate));
     if (ret != OPUS_OK) {
         av_log(avctx, AV_LOG_ERROR,
                "Failed to set bitrate: %s\n", opus_strerror(ret));
         return ret;
     }
 
     ret = opus_multistream_encoder_ctl(enc,
                                        OPUS_SET_COMPLEXITY(opts->complexity));
     if (ret != OPUS_OK)
         av_log(avctx, AV_LOG_WARNING,
                "Unable to set complexity: %s\n", opus_strerror(ret));
 
     ret = opus_multistream_encoder_ctl(enc, OPUS_SET_VBR(!!opts->vbr));
     if (ret != OPUS_OK)
         av_log(avctx, AV_LOG_WARNING,
                "Unable to set VBR: %s\n", opus_strerror(ret));
 
     ret = opus_multistream_encoder_ctl(enc,
                                        OPUS_SET_VBR_CONSTRAINT(opts->vbr == 2));
     if (ret != OPUS_OK)
         av_log(avctx, AV_LOG_WARNING,
                "Unable to set constrained VBR: %s\n", opus_strerror(ret));
 
     ret = opus_multistream_encoder_ctl(enc,
                                        OPUS_SET_PACKET_LOSS_PERC(opts->packet_loss));
     if (ret != OPUS_OK)
         av_log(avctx, AV_LOG_WARNING,
                "Unable to set expected packet loss percentage: %s\n",
                opus_strerror(ret));
 
     if (avctx->cutoff) {
         ret = opus_multistream_encoder_ctl(enc,
                                            OPUS_SET_MAX_BANDWIDTH(opts->max_bandwidth));
         if (ret != OPUS_OK)
             av_log(avctx, AV_LOG_WARNING,
                    "Unable to set maximum bandwidth: %s\n", opus_strerror(ret));
     }
 
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 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
     ret = opus_multistream_encoder_ctl(enc,
                                        OPUS_SET_PHASE_INVERSION_DISABLED(!opts->apply_phase_inv));
     if (ret != OPUS_OK)
         av_log(avctx, AV_LOG_WARNING,
                "Unable to set phase inversion: %s\n",
                opus_strerror(ret));
 #endif
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     return OPUS_OK;
 }
 
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 static int libopus_check_max_channels(AVCodecContext *avctx,
                                       int max_channels) {
     if (avctx->channels > max_channels) {
         av_log(avctx, AV_LOG_ERROR, "Opus mapping family undefined for %d channels.\n",
                avctx->channels);
         return AVERROR(EINVAL);
     }
 
     return 0;
 }
 
 static int libopus_check_vorbis_layout(AVCodecContext *avctx, int mapping_family) {
     av_assert2(avctx->channels < FF_ARRAY_ELEMS(ff_vorbis_channel_layouts));
 
     if (!avctx->channel_layout) {
         av_log(avctx, AV_LOG_WARNING,
                "No channel layout specified. Opus encoder will use Vorbis "
                "channel layout for %d channels.\n", avctx->channels);
     } else if (avctx->channel_layout != ff_vorbis_channel_layouts[avctx->channels - 1]) {
         char name[32];
         av_get_channel_layout_string(name, sizeof(name), avctx->channels,
                                      avctx->channel_layout);
         av_log(avctx, AV_LOG_ERROR,
                "Invalid channel layout %s for specified mapping family %d.\n",
                name, mapping_family);
 
         return AVERROR(EINVAL);
     }
 
     return 0;
 }
 
 static int libopus_validate_layout_and_get_channel_map(
         AVCodecContext *avctx,
         int mapping_family,
         const uint8_t ** channel_map_result)
 {
     const uint8_t * channel_map = NULL;
     int ret;
 
     switch (mapping_family) {
     case -1:
         ret = libopus_check_max_channels(avctx, 8);
         if (ret == 0) {
             ret = libopus_check_vorbis_layout(avctx, mapping_family);
             /* Channels do not need to be reordered. */
         }
 
         break;
     case 0:
         ret = libopus_check_max_channels(avctx, 2);
         if (ret == 0) {
             ret = libopus_check_vorbis_layout(avctx, mapping_family);
         }
         break;
     case 1:
         /* Opus expects channels to be in Vorbis order. */
         ret = libopus_check_max_channels(avctx, 8);
         if (ret == 0) {
             ret = libopus_check_vorbis_layout(avctx, mapping_family);
             channel_map = ff_vorbis_channel_layout_offsets[avctx->channels - 1];
         }
         break;
     case 255:
         ret = libopus_check_max_channels(avctx, 254);
         break;
     default:
         av_log(avctx, AV_LOG_WARNING,
                "Unknown channel mapping family %d. Output channel layout may be invalid.\n",
                mapping_family);
         ret = 0;
     }
 
     *channel_map_result = channel_map;
     return ret;
 }
 
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 static av_cold int libopus_encode_init(AVCodecContext *avctx)
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 {
     LibopusEncContext *opus = avctx->priv_data;
     OpusMSEncoder *enc;
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     uint8_t libopus_channel_mapping[255];
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     int ret = OPUS_OK;
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     int av_ret;
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     int coupled_stream_count, header_size, frame_size;
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     int mapping_family;
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     frame_size = opus->opts.frame_duration * 48000 / 1000;
     switch (frame_size) {
     case 120:
     case 240:
         if (opus->opts.application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)
             av_log(avctx, AV_LOG_WARNING,
                    "LPC mode cannot be used with a frame duration of less "
                    "than 10ms. Enabling restricted low-delay mode.\n"
                    "Use a longer frame duration if this is not what you want.\n");
         /* Frame sizes less than 10 ms can only use MDCT mode, so switching to
          * RESTRICTED_LOWDELAY avoids an unnecessary extra 2.5ms lookahead. */
         opus->opts.application = OPUS_APPLICATION_RESTRICTED_LOWDELAY;
     case 480:
     case 960:
     case 1920:
     case 2880:
         opus->opts.packet_size =
         avctx->frame_size      = frame_size * avctx->sample_rate / 48000;
         break;
     default:
         av_log(avctx, AV_LOG_ERROR, "Invalid frame duration: %g.\n"
                "Frame duration must be exactly one of: 2.5, 5, 10, 20, 40 or 60.\n",
                opus->opts.frame_duration);
         return AVERROR(EINVAL);
     }
 
     if (avctx->compression_level < 0 || avctx->compression_level > 10) {
         av_log(avctx, AV_LOG_WARNING,
                "Compression level must be in the range 0 to 10. "
                "Defaulting to 10.\n");
         opus->opts.complexity = 10;
     } else {
         opus->opts.complexity = avctx->compression_level;
     }
 
     if (avctx->cutoff) {
         switch (avctx->cutoff) {
         case  4000:
             opus->opts.max_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
             break;
         case  6000:
             opus->opts.max_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
             break;
         case  8000:
             opus->opts.max_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
             break;
         case 12000:
             opus->opts.max_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
             break;
         case 20000:
             opus->opts.max_bandwidth = OPUS_BANDWIDTH_FULLBAND;
             break;
         default:
             av_log(avctx, AV_LOG_WARNING,
                    "Invalid frequency cutoff: %d. Using default maximum bandwidth.\n"
                    "Cutoff frequency must be exactly one of: 4000, 6000, 8000, 12000 or 20000.\n",
                    avctx->cutoff);
             avctx->cutoff = 0;
         }
     }
 
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     /* Channels may need to be reordered to match opus mapping. */
     av_ret = libopus_validate_layout_and_get_channel_map(avctx, opus->opts.mapping_family,
                                                          &opus->encoder_channel_map);
     if (av_ret) {
         return av_ret;
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     }
 
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     if (opus->opts.mapping_family == -1) {
         /* By default, use mapping family 1 for the header but use the older
          * libopus multistream API to avoid surround masking. */
 
         /* Set the mapping family so that the value is correct in the header */
         mapping_family = avctx->channels > 2 ? 1 : 0;
         coupled_stream_count = opus_coupled_streams[avctx->channels - 1];
         opus->stream_count   = avctx->channels - coupled_stream_count;
         memcpy(libopus_channel_mapping,
                opus_vorbis_channel_map[avctx->channels - 1],
                avctx->channels * sizeof(*libopus_channel_mapping));
 
         enc = opus_multistream_encoder_create(
             avctx->sample_rate, avctx->channels, opus->stream_count,
             coupled_stream_count,
             libavcodec_libopus_channel_map[avctx->channels - 1],
             opus->opts.application, &ret);
     } else {
         /* Use the newer multistream API. The encoder will set the channel
          * mapping and coupled stream counts to its internal defaults and will
          * use surround masking analysis to save bits. */
         mapping_family = opus->opts.mapping_family;
         enc = opus_multistream_surround_encoder_create(
             avctx->sample_rate, avctx->channels, mapping_family,
             &opus->stream_count, &coupled_stream_count, libopus_channel_mapping,
             opus->opts.application, &ret);
     }
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     if (ret != OPUS_OK) {
         av_log(avctx, AV_LOG_ERROR,
                "Failed to create encoder: %s\n", opus_strerror(ret));
         return ff_opus_error_to_averror(ret);
     }
 
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     if (!avctx->bit_rate) {
         /* Sane default copied from opusenc */
         avctx->bit_rate = 64000 * opus->stream_count +
                           32000 * coupled_stream_count;
         av_log(avctx, AV_LOG_WARNING,
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                "No bit rate set. Defaulting to %"PRId64" bps.\n", avctx->bit_rate);
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     }
 
     if (avctx->bit_rate < 500 || avctx->bit_rate > 256000 * avctx->channels) {
         av_log(avctx, AV_LOG_ERROR, "The bit rate %"PRId64" bps is unsupported. "
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                "Please choose a value between 500 and %d.\n", avctx->bit_rate,
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                256000 * avctx->channels);
         ret = AVERROR(EINVAL);
         goto fail;
     }
 
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     ret = libopus_configure_encoder(avctx, enc, &opus->opts);
     if (ret != OPUS_OK) {
         ret = ff_opus_error_to_averror(ret);
         goto fail;
     }
 
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     /* Header includes channel mapping table if and only if mapping family is NOT 0 */
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     header_size = 19 + (mapping_family == 0 ? 0 : 2 + avctx->channels);
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     avctx->extradata = av_malloc(header_size + AV_INPUT_BUFFER_PADDING_SIZE);
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     if (!avctx->extradata) {
         av_log(avctx, AV_LOG_ERROR, "Failed to allocate extradata.\n");
         ret = AVERROR(ENOMEM);
         goto fail;
     }
     avctx->extradata_size = header_size;
 
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     opus->samples = av_mallocz_array(frame_size, avctx->channels *
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                                av_get_bytes_per_sample(avctx->sample_fmt));
     if (!opus->samples) {
         av_log(avctx, AV_LOG_ERROR, "Failed to allocate samples buffer.\n");
         ret = AVERROR(ENOMEM);
         goto fail;
     }
 
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     ret = opus_multistream_encoder_ctl(enc, OPUS_GET_LOOKAHEAD(&avctx->initial_padding));
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     if (ret != OPUS_OK)
         av_log(avctx, AV_LOG_WARNING,
                "Unable to get number of lookahead samples: %s\n",
                opus_strerror(ret));
 
     libopus_write_header(avctx, opus->stream_count, coupled_stream_count,
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                          mapping_family, libopus_channel_mapping);
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     ff_af_queue_init(avctx, &opus->afq);
 
     opus->enc = enc;
 
     return 0;
 
 fail:
     opus_multistream_encoder_destroy(enc);
     av_freep(&avctx->extradata);
     return ret;
 }
 
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 static void libopus_copy_samples_with_channel_map(
     uint8_t *dst, const uint8_t *src, const uint8_t *channel_map,
     int nb_channels, int nb_samples, int bytes_per_sample) {
     int sample, channel;
     for (sample = 0; sample < nb_samples; ++sample) {
         for (channel = 0; channel < nb_channels; ++channel) {
             const size_t src_pos = bytes_per_sample * (nb_channels * sample + channel);
             const size_t dst_pos = bytes_per_sample * (nb_channels * sample + channel_map[channel]);
 
             memcpy(&dst[dst_pos], &src[src_pos], bytes_per_sample);
         }
     }
 }
 
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 static int libopus_encode(AVCodecContext *avctx, AVPacket *avpkt,
                           const AVFrame *frame, int *got_packet_ptr)
 {
     LibopusEncContext *opus = avctx->priv_data;
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     const int bytes_per_sample = av_get_bytes_per_sample(avctx->sample_fmt);
     const int sample_size      = avctx->channels * bytes_per_sample;
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     uint8_t *audio;
     int ret;
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     int discard_padding;
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     if (frame) {
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         ret = ff_af_queue_add(&opus->afq, frame);
         if (ret < 0)
             return ret;
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         if (opus->encoder_channel_map != NULL) {
             audio = opus->samples;
             libopus_copy_samples_with_channel_map(
                 audio, frame->data[0], opus->encoder_channel_map,
                 avctx->channels, frame->nb_samples, bytes_per_sample);
         } else if (frame->nb_samples < opus->opts.packet_size) {
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             audio = opus->samples;
             memcpy(audio, frame->data[0], frame->nb_samples * sample_size);
         } else
             audio = frame->data[0];
     } else {
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         if (!opus->afq.remaining_samples || (!opus->afq.frame_alloc && !opus->afq.frame_count))
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             return 0;
         audio = opus->samples;
         memset(audio, 0, opus->opts.packet_size * sample_size);
     }
 
     /* Maximum packet size taken from opusenc in opus-tools. 60ms packets
      * consist of 3 frames in one packet. The maximum frame size is 1275
      * bytes along with the largest possible packet header of 7 bytes. */
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     if ((ret = ff_alloc_packet2(avctx, avpkt, (1275 * 3 + 7) * opus->stream_count, 0)) < 0)
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         return ret;
 
     if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
         ret = opus_multistream_encode_float(opus->enc, (float *)audio,
                                             opus->opts.packet_size,
                                             avpkt->data, avpkt->size);
     else
         ret = opus_multistream_encode(opus->enc, (opus_int16 *)audio,
                                       opus->opts.packet_size,
                                       avpkt->data, avpkt->size);
 
     if (ret < 0) {
         av_log(avctx, AV_LOG_ERROR,
                "Error encoding frame: %s\n", opus_strerror(ret));
         return ff_opus_error_to_averror(ret);
     }
 
     av_shrink_packet(avpkt, ret);
 
     ff_af_queue_remove(&opus->afq, opus->opts.packet_size,
                        &avpkt->pts, &avpkt->duration);
 
ae12d655
     discard_padding = opus->opts.packet_size - avpkt->duration;
     // Check if subtraction resulted in an overflow
     if ((discard_padding < opus->opts.packet_size) != (avpkt->duration > 0)) {
c2f861ca
         av_packet_unref(avpkt);
ae12d655
         av_free(avpkt);
         return AVERROR(EINVAL);
     }
     if (discard_padding > 0) {
         uint8_t* side_data = av_packet_new_side_data(avpkt,
                                                      AV_PKT_DATA_SKIP_SAMPLES,
                                                      10);
fb33bff9
         if(!side_data) {
c2f861ca
             av_packet_unref(avpkt);
ae12d655
             av_free(avpkt);
             return AVERROR(ENOMEM);
         }
         AV_WL32(side_data + 4, discard_padding);
     }
 
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     *got_packet_ptr = 1;
 
     return 0;
 }
 
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 static av_cold int libopus_encode_close(AVCodecContext *avctx)
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 {
     LibopusEncContext *opus = avctx->priv_data;
 
     opus_multistream_encoder_destroy(opus->enc);
 
     ff_af_queue_close(&opus->afq);
 
     av_freep(&opus->samples);
     av_freep(&avctx->extradata);
 
     return 0;
 }
 
 #define OFFSET(x) offsetof(LibopusEncContext, opts.x)
 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
 static const AVOption libopus_options[] = {
     { "application",    "Intended application type",           OFFSET(application),    AV_OPT_TYPE_INT,   { .i64 = OPUS_APPLICATION_AUDIO }, OPUS_APPLICATION_VOIP, OPUS_APPLICATION_RESTRICTED_LOWDELAY, FLAGS, "application" },
         { "voip",           "Favor improved speech intelligibility",   0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_VOIP },                0, 0, FLAGS, "application" },
         { "audio",          "Favor faithfulness to the input",         0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_AUDIO },               0, 0, FLAGS, "application" },
         { "lowdelay",       "Restrict to only the lowest delay modes", 0, AV_OPT_TYPE_CONST, { .i64 = OPUS_APPLICATION_RESTRICTED_LOWDELAY }, 0, 0, FLAGS, "application" },
74906d37
     { "frame_duration", "Duration of a frame in milliseconds", OFFSET(frame_duration), AV_OPT_TYPE_FLOAT, { .dbl = 20.0 }, 2.5, 60.0, FLAGS },
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     { "packet_loss",    "Expected packet loss percentage",     OFFSET(packet_loss),    AV_OPT_TYPE_INT,   { .i64 = 0 },    0,   100,  FLAGS },
     { "vbr",            "Variable bit rate mode",              OFFSET(vbr),            AV_OPT_TYPE_INT,   { .i64 = 1 },    0,   2,    FLAGS, "vbr" },
         { "off",            "Use constant bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 0 }, 0, 0, FLAGS, "vbr" },
         { "on",             "Use variable bit rate", 0, AV_OPT_TYPE_CONST, { .i64 = 1 }, 0, 0, FLAGS, "vbr" },
         { "constrained",    "Use constrained VBR",   0, AV_OPT_TYPE_CONST, { .i64 = 2 }, 0, 0, FLAGS, "vbr" },
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     { "mapping_family", "Channel Mapping Family",              OFFSET(mapping_family), AV_OPT_TYPE_INT,   { .i64 = -1 },   -1,  255,  FLAGS, "mapping_family" },
204c7caf
 #ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
     { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, FLAGS },
 #endif
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     { NULL },
 };
 
 static const AVClass libopus_class = {
     .class_name = "libopus",
     .item_name  = av_default_item_name,
     .option     = libopus_options,
     .version    = LIBAVUTIL_VERSION_INT,
 };
 
 static const AVCodecDefault libopus_defaults[] = {
     { "b",                 "0" },
     { "compression_level", "10" },
     { NULL },
 };
 
 static const int libopus_sample_rates[] = {
     48000, 24000, 16000, 12000, 8000, 0,
 };
 
 AVCodec ff_libopus_encoder = {
     .name            = "libopus",
b2bed932
     .long_name       = NULL_IF_CONFIG_SMALL("libopus Opus"),
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     .type            = AVMEDIA_TYPE_AUDIO,
     .id              = AV_CODEC_ID_OPUS,
     .priv_data_size  = sizeof(LibopusEncContext),
     .init            = libopus_encode_init,
     .encode2         = libopus_encode,
     .close           = libopus_encode_close,
def97856
     .capabilities    = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
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     .sample_fmts     = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
                                                       AV_SAMPLE_FMT_FLT,
                                                       AV_SAMPLE_FMT_NONE },
     .supported_samplerates = libopus_sample_rates,
     .priv_class      = &libopus_class,
     .defaults        = libopus_defaults,
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     .wrapper_name    = "libopus",
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 };